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  <changelog author="Adam Tkac &lt;atkac redhat com&gt; - 1.2.12-4" date="1356091200">- rebuild against new libjpeg</changelog>
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  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-6" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-7" date="1375444800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-8" date="1402056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-9" date="1408104000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Moez Roy &lt;moez.roy@gmail.com&gt; - 1.2.12-11" date="1414065600">- Add global harderning flags</changelog>
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  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-18" date="1362916800">- Add silence patch as we don't built with DJBFFT enabled</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.7.4-19" date="1409400000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
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  <changelog author="Nicolas Chauvet &lt;nicolas.chauvet@kwizart.fr&gt; - 0.7.4-21" date="1471953600">- Fix macro in comment</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;kwizart@rpmfusion.org&gt; - 0.7.4-22" date="1489838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-24" date="1490011200">- Multilibs support - rhbz#1433758
- Simplify description
- Convert AUTHORS to UTF-8
- Drop Groups
- Add Obsoletes/Provides for a52dec-devel</changelog>
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- Use sed instead of perl to avoid a build dependency</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-26" date="1490875200">- Fix multilibs transition</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-27" date="1491307200">- Fix others 3rd part repos</changelog>
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  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-18" date="1362916800">- Add silence patch as we don't built with DJBFFT enabled</changelog>
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  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-24" date="1490011200">- Multilibs support - rhbz#1433758
- Simplify description
- Convert AUTHORS to UTF-8
- Drop Groups
- Add Obsoletes/Provides for a52dec-devel</changelog>
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- Use sed instead of perl to avoid a build dependency</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.1-1" date="1504353600">- First version of this spec file</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
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  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
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  <version epoch="8" ver="1.5.10" rel="26.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-17" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 1.5.10-18" date="1349956800">- rebuild (audiofile)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-19" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Than Ngo &lt;than@redhat.com&gt; - 1.5.10-20" date="1362657600">- fix FTBFS in rawhide</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-21" date="1362830400">- unify KDE 3 autotools fixes between packages</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-22" date="1364299200">- use automake --force-missing to get aarch64 support (#925029/#925627)
- also use automake --copy (the default is symlinking)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-23" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-24" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 8:1.5.10-25" date="1407931200">- kill rpaths harder</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-26" date="1408104000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-17" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 1.5.10-18" date="1349956800">- rebuild (audiofile)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-19" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Than Ngo &lt;than@redhat.com&gt; - 1.5.10-20" date="1362657600">- fix FTBFS in rawhide</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-21" date="1362830400">- unify KDE 3 autotools fixes between packages</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-22" date="1364299200">- use automake --force-missing to get aarch64 support (#925029/#925627)
- also use automake --copy (the default is symlinking)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-23" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-24" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 8:1.5.10-25" date="1407931200">- kill rpaths harder</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-26" date="1408104000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
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  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-17" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 1.5.10-18" date="1349956800">- rebuild (audiofile)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-19" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Than Ngo &lt;than@redhat.com&gt; - 1.5.10-20" date="1362657600">- fix FTBFS in rawhide</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-21" date="1362830400">- unify KDE 3 autotools fixes between packages</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-22" date="1364299200">- use automake --force-missing to get aarch64 support (#925029/#925627)
- also use automake --copy (the default is symlinking)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-23" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-24" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 8:1.5.10-25" date="1407931200">- kill rpaths harder</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-26" date="1408104000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
</package>
<package pkgid="1691f86982b96fc0c89d1c2238e10d25494badd71048f0850b5552c6a22f493b" name="asio-devel" arch="x86_64">
  <version epoch="0" ver="1.10.8" rel="1.el7_9"/>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1.10.4-5" date="1437566400">- rebuild for Boost 1.58</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; -1.10.6-1" date="1437912000">- Update to 1.10.6 version</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.10.6-2" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.10.6-3" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.6-4" date="1451736000">- Move from define to global</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.6-5" date="1451736001">- Remove useless pieces of the spec
- Conform to more recent SPEC style
- Fix date in changelog that was giving warnings</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.10.6-6" date="1452772800">- Rebuilt for Boost 1.60</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.10.6-7" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.7-1" date="1473768000">- Update to 1.10.7</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.8-1" date="1481457600">- Update to 1.10.8</changelog>
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<package pkgid="096e3bb2217706c17a2ce1928ddce98b27f2d9e4c5f4808850215ebc87c92fee" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2ed9861ba3611e4af06d70161186894d3341bc93a5188efdb3a57b2cc26d8913" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c3a747e881735f454d66d1035290065cf3de882e2d3682833bebf16a9016daf9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bff8ec3cbf949c50df3b095bbd69d64ed66f5aead82ed95bd7edc2082c2d169d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="8fc76fa884297d983bd3017677e73d0f23391ea55c5fe5fd5f502e4b062076dd" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3f2a5d032a61801176245c661f7d5912bbe0ba90e66bcf0565fe6a70bedf457d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8d56d6eae4d62e05e48f15e7fbce9bb1f0ab2a0928e6ba975ae7f4442598943e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="387ed9adb54f91afc3a654a8dfb8685ccd1d1ebe3845100b8a3a4b35c99cac29" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="37f2eee68747b2af518aeac477f4864f21986e5836a773ad7026eee557da9287" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0e3e260f83c2aae061d607df6932c231e2a69cab3524ac77ae6506fb889349cb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c47832983fb26371c3b5258dd405afd81140964910e5d08ec416d569973c3e85" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d7eb8b5fe138654e9fe9fea2e4af8836946ea9a008d3c4206044122e264a7459" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="472e821d86f7f08b7e4791d8c2226368012cd63fe82a30f035036e73c6508c62" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="055324d6f9f664afc90b1489bd1d02e38bca0aca2f3d9aefdc908b296c72abf3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="616129af9cb3691bb4f7f93b9951a641df0c6f2753dad4a2518ea4f90a613343" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a619e3176e1c9760596c699d185b1f8e90ce014686d5226e2f1fb6d6794aae22" name="asterisk" arch="x86_64">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6ed1e56c13841564cc63010d775a0d7b10499676b7e72e59d90695167fb9cca4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.0.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 14.5.0-4" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="59c97f5d676d06ca607f44e077a6d3ec8ca870bffcc1735ccfc4928ac80a936f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.1.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2146a86192cd20574a71db6b92ad6ccc415937bf0df2bbe486098bc4be990bee" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.1.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3c85083dbc6b8431866b530e9c126e88455f8c0f6edba008512464235580d5d8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.1.2" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267200">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267201">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267202">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7672f343643b9da9ecb59e7aef484ab226d7b99b121d2b752d94da2ba93bee5a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.1.3" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267200">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267201">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1023d646f04a4bb5ec0e1b762deb5b561975dea150a7365ed778d324b559c540" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.1.4" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267200">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3f6524b1b9310fe8b82a1d98c21b29b933dbf1e963383b8acdbdc0a580dc1acd" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.1.5" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="81f1b1abb433a4ca42060b014748d11ad3c034dd9af1a4c63cbaee0a8a5325ba" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.2.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d5bb4340271f49dbe3bba09419c5aeb9e6c87a4e6a70316c604fd4e40797c0c3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.2.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7eba15db07414d6041b2d8ace863f8e292f0fc121a82020959050c67fb011301" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.2.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
</package>
<package pkgid="6d12ce1086ab2b309179afe7c6096f8cd01df50aa74e85b033d191416aee30ff" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.2.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d179c75295fad54b3eff55ae61092891147713af9a1eb5cfc2fb082562f2465d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758404">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758405">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e593e6b075c5996f065d8c96c629be295ea7d26ee25635acd682a63c599ad980" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758403">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758404">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="60f37be022b374bc71cee4849c203025e8377d3ffa4f65db55eb95042edc7c7a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758402">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758403">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7568fb7835ff9ea0cca875a96d8728ba3629415b948bda81c387ea6c66a297df" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758401">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0b195d1b743509a1b9b9d54b72994986cff11e9bb8aeecdf96b90ea7f901281e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c1403641d24b4e307ffe95df9ebb4a426c0b7ac7f7f0e36d8a8a7ad21e641fbe" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3e67161656499b91a895781741acabc9022498b48ad7b6de654e3f65e1a37726" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.6.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="45b681c08bc2cc471b829df28d81adefc427bb2475d97139102a49349aab8cd7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
</package>
<package pkgid="5eb9a6e68b92440f05cb6d39955583eb735d6cc847e9e15f1ec90b41cfbc90d0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ce7e31e84b6340a6fd06dbfbb2596bcd23c75161e351202b4ec652af53efef17" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.7.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9947634158da0371d1c5aa8a2800388d8c1bc76a406c354cc2dbbf217ff30505" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.7.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e3e10a6efc6147f17701493283b695f623680a34d8a3803070d562acc49cc1ef" name="asterisk" arch="x86_64">
  <version epoch="0" ver="15.7.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.4-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1,
- 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-004: Crash when negotiating for T.38 with a declined stream
- When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
- responds with a declined media stream a crash will then occur in Asterisk.
-
- AST-2019-005: Remote Crash Vulnerability in audio transcoding
- When audio frames are given to the audio transcoding support in Asterisk the
- number of samples are examined and as part of this a message is output to
- indicate that no samples are present. A change was done to suppress this
- message for a particular scenario in which the message was not relevant. This
- change assumed that information about the origin of a frame will always exist
- when in reality it may not.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4778b29ed57e792cd6d743b3c690540f4949b87c177ede264014cce65da1b25d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5d08aa5098280ccb5a528e28cac71dadc2cc839febc43878b62dd42d3487d551" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e5803aad754727263fdbc20ac4a41b7a7591b4097aee0a50d9ec101d8bed0740" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e6310ad39cd6f35f170047c80c94761b8d46a788845e864e92e0381038234d4c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2992dc1eede473b958383ac5aca1569d8e2e8448c2d1ba05e86ff5158efcfb1b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.10.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="197c00e09f367304388d7befdcd8e2563960b0bc31cd7f41947eca068ed5a0f6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.11.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3a857a2b31bd07b9264e2c10093ef65015cb6574b6254fd3074769121338a2fe" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.11.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2d99b4fae486abb83d18c002aff91989e7f529e8f4b5c49f21acc645fc3d0a60" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.12.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7e3809387a29d0db17a40c37ed28500f37fb23bb3a209b6502de0b2fa8d93548" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.13.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bf792eceb6a59805faa97208cd2c8e6e39be984b1ae54abbff5bc143cf91ca82" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9fb9b7cabf95b1f6b9ae4768033c990b1abbc5b2c31ba316b348d64eeb073ae2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="61af9945d95cdb2e4eabe76bd554627599af57c0d8fd2fd087a1e3fc4a388411" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="56d2d9665438a82f4cde9ce16a9cda29a8dc1af9fff15888ddbf423dc87947ec" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b0ddef53479003fa0e491c109d8ff4a79a714c6c9400843741d1155c1c9f177e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="56b9295a61516fd9df7d0a98ae6e9b41dad03c61f80a50f9c37b8d3fe5fc849e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.16.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6580d0718c4b4ba54a54c391df75930b6dc91b40f581632efc58a2a8b46af7a7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.16.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="aefc5bed6429dbb254e21f48527d84f9e41232544a3505c70a0fe96da09b0525" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="113c5197da4d23c567239247daa9545e56ce9b05780ff2690bc368278568af3f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="79a931813750f2a8db3ffdb635970254f2fc8251325499d1114fdf73f16bd7c9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1720c6d99e5b06d57e5af328c1d9743266db4689869813f9e18553f71eca4acc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d84f7d812af21acd24f2267f485ef14482ee57b4c69b0d6b64028da62ab518db" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="99fece90ce5579d6cb94bf65a7e1b082224bfdc1d522cbb7150dd832299047dc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="c17136c6504a1102d5a056d617f18f2b74a3f5ea23cf1dadafbbf2e111cfa4c8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3ece1ef53ecfd707e953ace1b5787fd8ccabc453ae9633efe1c8d13cbaf2e076" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6eaf7abe45959f8ece6c70b471af186403edbd7fa88289505076aa9e7c1a058f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6dbb357b5565f5ece5e3d07455ae2ce946643d309bf76244ca36b7132b6fdd51" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="598bb5e87053b16201b729dc6e133d3c79e6d5d9641fd0df6431b65848aad786" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2286c304983a284ee996cf8c208ef13e176d885e31e16b9366bd63560cb7555f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c02f2acb68de296f678192f185b563f8a68071da57515dbd7cf58e4ea46150c8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dbcae22f1a9c70aae1aecd5720229ce8ad6e5f0af95f03b9283091f8d4562ed8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
</package>
<package pkgid="0204f7e6e7848806936cd7463e9b7773cafd2071f2d719a05c0f9fcb2d1f35e8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d624f169e3559d759e60bab02a4c832b31ae494f5efce542d4aa624e8eeaf459" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ddbf2752f0bab6481803e35c4de36a48685e81433f1b1a5ec57a202a48c53b5a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ecfc54098beef77271988cf82092c20ab1290a8141c27468ad6ada2b35bb0541" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9540b85a9b1284f394b8aff5a95e8a50c1a31fa10f1035ed1f89ca9a6c4e2d27" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="84f61f6993ab7571eb360f36be226477dff80f2a3165f4a9592021d869cf5c67" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2c8ca8e9ead7a775bf01fcfe41335b1c7f0190d3a2c4937ba54daaff860eea14" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="91c63e16e3945e631b743bed719d178eb55148d49a9102a77e35eb484cb55242" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5ef0673d5e5d9436c7a0c0ed0b1996a69b98537ac50be23a0267dd73f74710ca" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.27.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="de48b879564644d63e6681233554db4af6131a5d10913aaf2f68af79b3deac65" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="153cc8ae1c8585f88cc2532217abd7ec8d2c967b2da7b6e5378ed3480b645832" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="9b3571ff4a0e3af43b43b336f6c09bc80096f99a9195dff789e68803e002efa4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="dcf4733a8a279fce1e67b920eb5962e3afe6a423ff1f497f9dcb813f1dca71dc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.29.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9d778a75376ff27fdb7b71f31501ec859751fe9dfbdafd5c01d76351b1925868" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.29.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1889013222c6dcbe48047c7d86a729b7b8299cd213594e33e7fc9d6c9c0d69b7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e1bb2500752928b8a30eecdaad9fbc4562a4174b007f2edfc130b5aede172a18" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.30.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4f84928f6449683e58295c7fff272ca0874eed13ebf5af8bafeacac1e09b0c36" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.30.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 16.30.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/16.30.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 16.30.1
- ==============================
-
- Links:
- ------
-
-  [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.30.1.md)
-  [GitHub Diff](https://github.com/asterisk/asterisk/compare/16.30.0...16.30.1)
-  [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16.30.1.tar.gz)
-  [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
-  apply_patches: Use globbing instead of file/sort.
-  bundled_pjproject: Backport 2 SSL patches from upstream
-  bundled_pjproject: Backport security fixes from pjproject 2.13.1
-  apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
-              before applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="134d33a24b0eda27e732a62833355ca920d0b903684f90fe27b673b7807ab144" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e5bca3d99e3780c30222d322ffb06946263494eb068bfdcf3b00fcf03f1b8758" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="50fcf9a44c2a02c1ba13776df7fb17cb920ac90b854fd163b9b7bc15ae8ee8c1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8d189fd58974565ed30c311a41a2077c4e5551b00bf6707af9678020d65ac2c7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="de1789b49b2918ab7078d28c55caacddd664eea10fd21a8af45e807057fe4744" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4a3b5584c613c48a661be08f596cf55b2efdae2483bbfbac85c7a5085e0879a0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
</package>
<package pkgid="fd92329f404105d48313e14d967d2b32eb6787c6751de68ec0d9c9c2d0c6c780" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-1" date="1571227200">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="349c49ea9999c38b3736ab2d2468a15c023a985af6eaa97354abf24e55bddd24" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.6.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-1" date="1571227200">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-2" date="1571832000">- Added res_json module</changelog>
</package>
<package pkgid="41ad4fe75f1dbddc89584537e5f29166449443bf8d2a85aeb738c15140c34e66" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.6.2" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9e78d717a07524d085d344ddb506bef23bbdcdd298ee80b4e56354450b430869" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.7.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9b091cbe22ac06510b0ff6b2efa320fe094195541caed850b8c4ad1f80cf0dca" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.8.0" rel="1.el7_9"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a614a428e00031fc32917868848bf02006b31b774f1b3265a7c866869df1cda6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="16.9.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl) 
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes) 
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b3aab7050e7528d0c0887ca5db5f6ea4c509635a6c5f505e4a3ee547d20a2f48" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.0.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0d3fdc0df55ac075ab9d4fd1d86936d2149050eb22956e1158840710d5f3e184" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.0.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="48e805e727792460b431130b37e023bf0b6618235ba171e06b763fca04f02328" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.1.0" rel="1.el7_9"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1e9e4fb0a99cb29fb748dbd0f94e7095308f0c0450a72f5543262137b5e004d2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.2.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="409c1871ff450948709a696930185615b118574c33a38467b93ae6d6be561b58" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.3.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="65993589cd74e29b14bf4ceebe81b97dcb2d6b311ba66d9412711e0d210067b6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.4.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="12db36b4db0250df88af321c2c4724bbdb6a4d2fa32ff45037ef4d7ec69cd311" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.5.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0011414a15b9cf6c39016adc30e846568b7134effa4393958df5e01efc618f2b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.5.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c7bfa73c9f73ce7950921618bfeb980e47e0cbbac60b61ff8adbe50419e7df33" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.6.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f8796ebb17dcf14a479d8a720df94f4b0da3d7e161f4d38a7b73372f54ea26fc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="841ff7f0be916795af2e83fdb28a00b26cf4311305323eb34d6d1c26594077a7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="294e37d58151275aae79a9d6bc103d9b2f10a870d9617f7626edcb2eb0b5a48e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cc3adb29bb975e6ee3e12db7d2ce2776b93d2c23c5ba066efdd43731db909483" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5aa94e9f63e116a9974e3df0a94130df94203acf85f3f620463435776d90511f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="374ca590abdc8f6c75cfbc285ea75719cd917e9a248a3146144d461a8bff9010" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1923a954e1d63f483fe9da9816c3ff2ad5ede425ea02d9c1bddc003352fabce0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ff62ee26fea8f5f6ac58d5f5dbabcbd371de5b6df8e561568aff95dbb11bbd04" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6c7f4e76aaa768e515f0e240bb3714a934ee9f7b8c15234b1711ccec965327f8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="93662bc4cd9d5eb6df47eb9a18af3f3e15fada0d6db80f2eeb844ff00651e1c3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="9da9f8912db12f7084badb7c0d6fb2a1d0e8e1263cbe4c0d83fd0f10a7d53b01" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="21fd50c81cda41d98d3566ef637d5a9d1da313ef35fac33be17ea6f5623873c4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a2c7c5a2edb675a5b132e322553ef957b2b775d6f0e15f141f9cb3cf7bdf1ad5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3338d149f3c88b8dce259fdb404ab58321729056cb8e738be6ef290db1c91482" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="af28aa52a168ffb94b3659ad54f21280f43f937b398d4796101beabdf6c9ac3c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.10.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="486071df763123cf55d707028204265961093c1a456a5fbf7c7cd46f79f5ec2a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.10.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="955410388eb7e5722cf79d43c4c8b27c20b6705eb76178f01627a2dfedaae732" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4e798a1ab0a28b03cae68a8ff736fe23d7c5a0cd691d7157ff2321c1c5c3d0a0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b6ef17ee11557bae21198498f13728dbd9788b5252826eeb9f8a27b120a62f2c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="bd5b4518be7eda89dadf959e98531d012a9055aaabc6e6e52da200ceeab031a5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e52cf83780825b2f1d60a6155e9461b1ef5c50d79e5ec3bbba467605180f52fb" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.11.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fd438df30239884af6a41c87d42f7b8ed1a8bba241ad7f500ea23c2758da9d98" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="63f89cb3585c5379350ac16862b8a0fa0d7539f884298c505ad1d01112fda91b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.12.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="755118054700ab2c7b4c0dcac401df4e5302ca85a7b033f40b8f0078d75f5bba" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c8d960f4edb6ece5326c39786a557fb1532ec92a3facd97ef851974aae2b4607" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b74cf8053fe0f2950248e1695376ccb1a676f69b56ec903e7b83ce53d55636c8" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="6020006aeb7865dd94da83460566f7f49d8383c035bf30686261492b8cde165c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="06b3ef27f3d4dd6d72b45f56fe052f1c234bbf256367c4e1aea57e28b7d3cf2f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="584394f35bb607019ee83c9be0256396e24a9d90fbf7ef352687fe97c25ebb5b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1e8757658f8d0dc5147f4b94bd288a97b8199d4a391ef21878555702ac2cb56a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e9e7a270f29b780e8568870524986c0b05d004d06f1ba2675d2ebbd25e3c570b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6a3a7a1fcb083ae3ce240bace1d86c32d2600737dd384dd9765eeae14d2f95ba" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.17.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9569b1a87487c22ccf411fed9050088584d46bea24e8d684b947549ae1583e78" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="d24dff774ddf9c55a901f75ebdbf6f3dd1cd66bdea4f359f5b22d8c41a21147b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.18.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="0309469e03076dce901397f3341d248ca2c4742c7f41e81494b3886229194e99" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="0225075be03ffc9bec96054ed3de0394ea11f62ea1a7f174e58c204bf1f37f50" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5318920c65b4f007bb7e4d01b77ff5f24d033ed682c328241755869df2d2261b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5784be1cde96504f507fa3395041e943ba6a7bdfa61cb44bc8d356d319289237" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.2.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="57f23caf017240a75c078a652dba404480ba58be58eb8f7a293e937e48f8f8da" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="224c96ffc3de3caa8a8d4d5fe8c25bcdac6aad2adc5f6bfb990c69533233d2e0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.20.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="6e85719802d5e9a3ee21a963b5f6b03207e06670c7c62f2ea300e303c62c33fe" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.20.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="f759682353c57136d8fb6b0d99de43e44b0d1fcb6f99ca8a583468339efe0ac7" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="674eadbf7241db157908537b52362bf65f766a7a460c5b265d5bf61a8d046cc6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="7ddb17afadd600894de3b67382880d5f0a5fff86e2f112bbd2d37fe4802e2804" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="dc0111f92d2db9f08d9175b6342a3ca263f0dafb4c423c4201934f6150d3003f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="9b80cfe7c6eb7e6c730ae5d28faa3312d00560eb435a001d46b206e149924472" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="6e5e89173849dc446ca76665548ff5187b1b11c3cb52f588e15f00ba2887ea81" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="6637343251fa8f6129b17b11ac53690eae81a660891bbaff9b6b880260ac7ac6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="e590b62f872152174ace5af89fccf47d73d7e466d7a591afcf4c709d6b6f767b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.24.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="6420705de1a7ff0d4a9037b2fc628eb32dd2a25742f0b0b8307c79b3b0bba439" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="916d524b0f5a5a1326e13404dbcd4d8ccb21a8295abd04338e574cc320a4754e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="1c7576caad7ba8c141e894345af3305192d215c464d95eb33e39ea345cc4b221" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="00e1447f6d58cf14f65461872c5e07bf9ce4125f2c97694380f09918f82752c6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.26.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-
- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
-  asterisk.c: Add option to restrict shell access from remote consoles.
-  res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="ec5833a757bcc9c6738cb92f21d6f796eecdb151f826776529e52de92875652c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="287e2489e6c034146922c0d1dc8aea4ecb4c3713c138e8ca537dc31a71e3fd93" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fad1e83859448d93bbe58fb21e67cd3aa2b40a173cf98abb5478fbfde3cd8fde" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e941fe45e4b2b782e4ed6c740e67b3c8c421637061a17b5ec1fe0e99b39fd364" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c2fd106256956a8621934b3ae1c76f32fe100b37c7c190d2c2d24d6faeaa819e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="2ea1b230a49faa3745c7c5f63db822028c720b4e47210969b55e60b8f7a66bf4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="03e91643afde53b059c6c9ddf3d841b94f466d34a83f8f9c3e7fedfef989ea47" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="93f1781b40510a549ce97108f1781640f3b5806e8cdce18a9e9eed0ad5bfb987" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="d8584507d460e2eedfcfdedf71673a175108b86ebec1df11d3b80f3dddc675c4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="87be0aabd59ce7cada61c8146a446ab4c2f4fa0b8134278754c437e309d6959c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e718f3fa928fb058d5e15bf7918c7a881244d705b53f36ee4fcc2df1f8b4d739" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="47a6e655961d59be87bb9764b09d3771baff0d200d46abf39d72e76fdc4280fe" name="asterisk" arch="x86_64">
  <version epoch="0" ver="18.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="976923e039e5709b53a104cb011c50cf8222cceb72deb7dc7b885068789334af" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="94e7475b8a840583710c5dccb3d2eec86ba39eb1c76e2c8952b5e8f23392926a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6a6acc7e5cf0a32194b44d20dbefc094c68e2567b7d9f0362659efe6ca049a75" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5303b5f9129d494384e6cdae2d70d5698e4017cfc6e929edfd912d3148c59a3b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="461a5bdef1df98ef9592064366644afb1b53dbf7727c16da90872de780e6b66d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7580eea183f41bf4da6131c20ea44b2122d76a9e249c71b3d48ee9cca0ebe21a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a6e7e7f6ba869c61fbb958c279f40c1c51328330ac46cadc6494af3f815a7e23" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="0c2c2565fe6728db790cf40a50aaae2d8dc3562d13b8380001dd00b7b5a69e9a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e340a9397d7a1ee89947fb2aaf1d6228286dd57dc3f7565888d3edd4ee28ac7b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.3.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9c9a8b12f66fd10e6214fd3f7ee280e18d53d3040cc011345b9e133c260a65ce" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0309f3893a5628f57aea587db5ea294718b387366a2e486159ebd424d791e300" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b25815a179d0e5897dd187182031e684a6e9462ac779acf53e69212d1c05e9b1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="145b83e4b776751c16dd895ecbd7e3408d9a935879418f4a2f89c56073e68921" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f433b995d6ecb83f98f708003ab9ecbc167dbe44aa3d22c5009eec3cc3ba800d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="7a74e87a3499fd0f0db333776ba24a0b1f28ae67708ad0590f4dc2052ba4a90e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="16581c0be918b2686e59a1aed7ef6f16eb63b162c184569254a72f1b45d9d2c0" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="217872c985c7864f48fe2e736b83264b06f703d3676737a7fc666bd40d6bf564" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2b662e17e6ef6cc8ec373bb87fbcbb9d21ef663989ba76ab9b526b92676eda56" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="81b7b91c0b87b676d22dfd9783c012d9004c0def954ee0b406a8518cb9c6897d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="19.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="cfdf7e248c78d764beb63cdd3940d89c928d4247a46c96bcd3a69685fa48f68c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="39fadab51978e369ca51a607078c64385ce8e3690d8fe4a4df50f77ce2d6ecb2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="713da2320c7edc39ccaf9e673d0183821f6549299383a43be2d733f096410dc6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9e2b83e5445b9b6e9d50383cfbab7966b97fb5e33d1b836e44b0af90e73e1f9f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="d7fd3d5864e6c289a8aef4d7863b254f66c560b2b219cfbc03d03f2847d92a0d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="15483a92fb975e50a39b5ab3471a9484bdb13492fc21382f091d1c31a33937a2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="c1daad61f4e87edf5505e6c20036126bf5c4c994e6e132f9a6e66b896f14a29c" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="b8e2aa4885c3c0697b790e61f3336185719074939be035734d63431d70acf5aa" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="7c135312a63dda161250815d998c94566bf4898fa348906dfed9159837e3c771" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="71ebf1aac17897c815413189e7309cbabcba145b0bce49fa8d3200a9e164193a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Upgrade Notes:
- --------------
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author:   George Joseph
-             Date:     2025-05-19
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-                       has been added that, when set, will prevent remote
-                       consoles from executing shell commands using the '!'
-                       prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author:   George Joseph
-                        Date:     2025-03-24
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="e1c16bf1ebb5573027aceec16445c5423d388027040ee29e1f83f9161c0247fc" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2adac532cc97a1b96bac51de85bd7348c4e9ece683fc77a5af8618afa63a36ef" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ccf0045acaf8269b7a5bb9f509be9a0b9fb248db1a4b583bc076249b477451f6" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="db04590d99ad5953883cef625df93a7c9ea101dfe286d0505296c8f5b8b69d70" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="d852315b3a8fcfd2f763f682111ae02eaf577b8146cb3a0a4125d38083246c01" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="53dd71e2214561be77b54eadf9bba12a57529408402793d06457e9c95e1cd780" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="89624f824972807f1dcd3b519f877a407720fb07ee35373ff63562283e7c3e1d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="72aabbe7c6fe8923dee0c50d350c4715b1fff0f0e48c904cbed9901a53e22cee" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="e8667fcb18433e7b28aaaac3c4e784b57d5f387c9a1868650c0811f4bab80751" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="f0268150c907f4e3b07a2473f7320d29e138e371b7e1a43f60b7c41ee1e2cf41" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="2333f7bcc2201e3ab1abf354d5832fd973d067fa78c9a40daee35a396ff0832e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="55fb710ba6f93786bd1c4b7f9ec5f0e8b8b513e26c8160eee46f4503bbccd8c1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="8fffb2620f214ff5e508779fa2cbb45ecad4026f8e092b4b81c7a20b5fd2626d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="4fdf3256d4d4b42bd60703cb37bdce25013826552f540566ba2ef0351eec331b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="b0e1d57ce5358edbbe09bd34b0010e63c93b6c1d9ec8a28a81b20d9841c58eb4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="346878784dc7eb809cccf429f5723e08c2866fd65e77133cdbae7dbb90c15039" name="asterisk" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="37899108538f50aae06b91ae77f2f93a6e13751d15afd296f6cd4d2e0289d2a3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="b28fded321ed54709fc8ada2dbadfce538cce68310e17ed91612b72adb361b9d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="484a6e12d3e69ad360b04e9b6f75b6cc923227a75f0b028a00ba4ad5455e81b3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="a28eebf733dcdd19e2445e4f0de29f62aa55c685da202bf56a0d39b5e05aabf3" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="fd4f1d34c46ff42914b86232c0989c7a040a0c33947c89ee5b14c3969fad5dd5" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="9e5e5b040337ccfdd910db89063a4cc4c824dfc31d613a8ec9e7c15c507d32d2" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="fe456946d1e8b8008f7a2f242ea21967f19535c5dd8e5c25bf5919d623887b83" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="86da53580413c56fd4afb06b4ca1234d418897c3420d4b48e982e3df32a4de3b" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="db523343b9e4d4357668ccf41421b5acc731046a90e2d75a3c66a217bea9d1c1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="f3c25a728c8edfcde373a2df7268ae13712d1e6b1aae65135ca3edf9850f2916" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="03d31b94b5b7c116d201826cb88bdbb1bf62580ffe60faf46419c450e1e8193d" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="59153e91f209af6857f4bfc76f316d0bc523161fd42a5ba59f8f924fd886d4db" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="2d563a4e6b3d46851745c55a35016b677433fe1e36e4f3d786bde11a9eb6572e" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="db3f4111a8c193bf665839c4c99cb16bd431126aaa8bef5a68fa55d8d52445f4" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="859c455b2ca12a7d49a6f46c4266a417698bc4c0c7d6cb1ca3f559f720f317d9" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="1cb0c35773bb8227595cd0ce88cd7331e28311a505c9eaca9fea48ad3a15869a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="1b4c0008f674c24038ccd7069052e0cae69ffe72a03b2cf05ddc8f57d081bddd" name="asterisk" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="276614cea7796592ff5a5b1712ce870acb02ea1b59b9ef4516e379cbc7de2b49" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="d2657c2b5bbf85480369c0e8dee86bace0e9e5c5350b081f6737b3f0bef45443" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="080c55dbdfb59b1fbb49e673ce5043ad70538fa23b8a21ce59c5030e7c10e69f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="2de5d1048efe876956f21768d9a4406a42aa97cc9c818a3db4598c81a19404d1" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="4f1eaa1b53a2402cc11a763482ff2f5ee7fa659758d7e1d9e453cfd2117a308f" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="bce676ee83bbd048a05fb0fdd3a5dc43d81fafeba5f548eb3ffe1889abf3f02a" name="asterisk" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="fb356cbd9987d63f71f9995a0e44e9339b3272ef519240390c8a7e5b0e4f6561" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="14818d82eced4d9bbd2a0874bb5fedf53cb7715cfc2e8c1ae2358c46a23798ff" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0b0f5ea414aa5bdd912f6524c0b6de36fc7036b0448e5b933b2d5d0da68be790" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b3e0e6e37fa77e615b8fe75fb87fd8c9c88aacf7a9af9641fd8c46e335560d71" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="8bf63921ed3f69e5a0f36eb8618a7bff96099325a5036e77921d873b197169e3" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0c5c950c39edcc72881dd82d565071d375bc2590341695a552800ec6d452ed6d" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="301ec12282062e578b9eac82f6d18bf34581d345d5acabe55d2cbf95be318bb0" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a8bcda3acc892f26d0ee4637b2d06c7f11f51cf332474fa78d905b8f4492415d" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="6a36f354bdd742072a7571832587120ec706e55a07b9655293b44e96f6c16609" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7a24f472d2bf2fbe094032a870d499e852898e8dcb9a27f04f48924c30cc5090" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="82dd62df10cd1cc1a8fa24ee813bacd003f3c591fce06957fe0d970dc3d36973" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ffd7ef86b8a6d84e1755ce13832abd14a8de2d1f3a90554b8d8a9f8fd17550e6" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8f12df9865e84605c18d5f77e3a29ecbd0efbba03508360894cbb2b5b2de0ab1" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="764bc64861b86d65a740de7702714134f59dd94791ed05a60c61dc05f27a02a5" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="e61f125512f356790c303d4085364a61c73a62c295a0be43458ce2923b5fbcc0" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5970bbe3d247821a6600f798c05e207594b23db440620d086e1e498795b3cd30" name="asterisk-addons" arch="x86_64">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="86b26f1f318e4b070b4fe4c960e327818944e128325931f390ba128ad5bb7777" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e252736f905778502b9bdcb0f5661e46db75ed01a439b0595d3966b0510343f8" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6dce5d03f3debc71c8725831b60ba752d7ff5a7fe39f3e32b1af35c60cc8095a" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d042dfac9c8dadec34856f748ca3ecaf76da06a52a4e4a9eedf1b4eec70d5b71" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="aea12112d4cb08b23275805080a0906eeef631329e2b54dcc517041352dd121b" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d3d665c847406417f1ef72cf6fd2f7fd3b3de5eafc3633d4fe1c919af7128cc2" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9e49e86a7cb0243896b1b399ab106ec60a6c391cdb0b62bdc4a91907e3c308c9" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dc3abfaf6ef795c873372ef33f6132bed57b0c666995199aedba4d82c6784ef6" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="e2b438c993f55d3c9355af66cffa3a44330276c0762d1aca692689835a01f2a1" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="97fab3ed034b37c3f54509ce38b18432a15a4026a69bba79b8b7763ec5a0585e" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e7dca495d25709efa2dae067d25a8a2e398b7c44775f1ef45601b2b88c8401dd" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1e6a096d6cd929599167d3b8f66e72f17c7d6f2e1dd7342f5e6aa606a514a058" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bb60ccad953c4ff98436659a9da2e1e59c3fe940d286f116a0febab3960d54bb" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="45d13ec4f950c97096b30853dabf0d5e8cbb6cc51bad12cb6a99e12704c3c8bf" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="b27c65bb4c4589d15c41d2a7b0868dca5bfd68bcc83c23d14d5d91b4e71d66b8" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="63150f4c8b3d46bdbc7d4456312cd75dc9d5d4438b74ec415da177786f891513" name="asterisk-addons-bluetooth" arch="x86_64">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9ef8071cb06a5dd915044e92897c12fd40ab6e81e4afdb83798ad026bce7aa0c" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9b0b7c9ba995304abe7eeb7b41d4677453d3a2cb0555e51d76a613ef4f72dfac" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ec17180a6c1450bf02b89cbabef3b4566fa6cca64922ec69f6aae9220805bd62" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b9061a3fdc4b1adfa75c0f07f1acd970cfe5a0d969c58ade026ebad8b6f61f33" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="4518b8ccf5c80b1d0428c897d8eed9a29e5bc8be196ce42380868db3debdc54c" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="51b89941c899c37e4cbce14fc2133052c9cbbb3e6527f108e6103f47bf5f96bd" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b0aff2aed158b6f2a61178343a373317bf5b2532dae4979aa3d679b81b4acfbe" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0d96f8a1fb9d4ea0d0e194c4e233aa4da21526c366dc886d5a812f9c698ddc73" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="66ab181f872c06220fe08562b3c5ddcb78e00417d79fdc177dfe9e10b3ffdbf5" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8219a428e5e09777425e35cd79eced84373098ae99350d525a002027bff19ef0" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4baefca14ec02acdbc2083f8caab1fe5c52575651d20e3dc28a8f0de28639761" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fb07dd8c0690b1248df58c8894063461ae74c9a1c2f30cc73c4724f1d44d6955" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0e0ab5387b78c7850d4a36c2e3d565d352e6003204817f3c70eaf712fdae6b2b" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="7688aaa8f7d968a8ca3818f097b8810681dfe8f5f59c7a6d663e8f2c07479af3" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="ee16831f545bbea0e9d69bd2eb169dcd8213943c649ca391f26bdd932d2fcdc6" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a4b4cc7b4d292d04b6009f2f3e185a5abdc296a613916f75ed22fa9cdbd0630b" name="asterisk-addons-core" arch="x86_64">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="46f3750a2cc899b2b9767f975f1004cd48d9817a2354f05337d392c07646ad6d" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="36620dd89f93663bf981176ef5e6f0c62108b32831d55f9d37b5fc013b1fc02f" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0a8b04b44e7da7c373b7164089975f69e68dc59a7c2fc8d335117e5ec803b857" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4d3305480829feeb44c9926d5089ffc93aa0c0c316675c49ac8bf784449f4054" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="33e8c5268a9a083a5f000ddb84f587a34a092a037080c4af6b35ffbb2078a464" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c80e7cd7ea6c814cf9af50efd511968960b9e2abf0fdc97abaeed50aa00319b7" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="29736dcd9c5d8980a07cff318bd7f8bb8fb2beca93f53cd0bb44e73afe9d3408" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="709b9c0433538ba6cd27ea8b173fc9abbe9473bdebceeb678bbbec3651720234" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="216c722d851fff68a002563bd4c1a666fc323af264a7fb8347a1f116d327525a" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0776c5666c0b93de666bbe8bda148b75927b7806642004c966a4c53f25d342bb" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2c7ed0123a491acdbf2475a84e60b6a51f6094473cdfbe5744c5936f0a23ea15" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="738c0db59ba225b313359e39192ef957698dd75656254d729a1e1271b4797748" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a9644190462748663234bd418a45920b9fa5bd014d0d45b54dd966e460651fc1" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="fe974946a09ce6fdaa1356349549a335ebd6f9b2943a88746f1d37e907e3eff9" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="8be7ee3fbb9165316b9c495f773bdbe4519fe88796bb6532cd54d410ea9b625c" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6f99d0a392c56ea471056ab91e2c97776b5d4bdb5505a7578b2942a47b17ea30" name="asterisk-addons-fax" arch="x86_64">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="62af7f98786a63c039582872446f5ba1cbea1fa3f3b49686e2329eb2015c7a78" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="351c1da1e7a315b5831ba96bf3805ee454b22e59075359d04fc1e703c0d46616" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7c9a1e08dfd769627daa22d8cd0724c042149a0de0868a2c701e5558514b462d" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="684d3582fb33afe11648dcaa60e974c04794976a0ed77865932eb069d335e7d0" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="d1fb2868f070ae7cb0319ec69326e1663190edf96d2c7574981d726205ae5b83" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c267776d82e03e3a82dd6756b3c60d6789c6f398251664d78fe7812ef6dc95a9" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fc4055ff8a688fa5858ff1b6dbad74d1593caf307e982c06721cf3e0a0b4e0d9" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5b394beaafcc152a53abda5c427295a5e246c57ca1d6c02f93f8deaf935df170" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="9676ffae7fe7df8c9c9f6ba74412b26b6497b118d2ba2efc32d6b25494791d04" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f1b47470b06fec289d04dd0a285b1abefc216207bb25281d4511a26b46cd94f1" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f02de713cfac828716317f9d4bc8e7900f696cad405fa5f6ec562d919f3b4bd8" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="43ce01ec65d63280d7a569ed72ef099eb1321fb3cd0d791bfdf62dcd452fa200" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bfa725d464c0451d7728d473a60299d58d80f4df96801f630921aad8514cb727" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="f8cd611d419a6e23826c078775516b466f21d04ebf3c809ce4937c8382e6aff4" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="385f5f7fddaf1605b12032c398e976da8d138c9f2512c8a4cdfcd7aa58a3dbe5" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f159e1daed9216f1e0a32023521802d44bbfe9431e19e57cd6d2a64e3309684a" name="asterisk-addons-mysql" arch="x86_64">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="88882d371b7bee38ccbe09f04fc088eb64ed0fdb4e7c01d034e5f3138e51a480" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="398e211d8fcf83d2a49e7e890f448ccbf7fe6d4a2292c65b29219f67c60155ee" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ac06709e8991625098ff4bcf491418886e4ca536c3302b91410aed8dc2bbddf4" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="025f27d7556e81ffd696b5d8357c1a42d14420006a46b16ee586d1672f620f16" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="64faba2b3ed5d7ef064ae0bb0b1ee805f8fd14816c414fbc5d6c3e00f742b034" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="997685bb0a6b2026e4182541fb284fa4c43195277b81301c65d319da8adf8ca2" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d38371261205506e84047f7022c070d79abbea5df30854319b210547f1f4046e" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ba38381322f6e6e547229499a534739157c01b715373d271e4616903768cec73" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="81bed4e8ac71da601daedfc403fca5b15ca5a82fffc3385db795f6740e764e00" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cc79bb0698309b62f73bd026a84fa902d7274f9e99d03858e55f8fb938c71723" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2291642550991ade284143fe7221d737726c27850b3e066751314941fc79331f" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0d80823016aa0642f1ae1d5b8bd5953a2e62781cfd23168efc3983752394c166" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e2c1888e45c350f81b90b9db6c0f0d8939997b406b39538fc9eea7268e7c6ff2" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="25043384562f5b5362a23871d8405c7a870dc0f55f43c8622b14b809e64369cc" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="41a649ea10f2056dc717a6e1ee96a7ce3a4856acbf809c54dbb31cfdbdb231a3" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7c190eba6a1cd48802ea9ebd195728b495b321a1752294b56a2b4dd6444016fe" name="asterisk-addons-ooh323" arch="x86_64">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="23beb56fa45effa9d04c2a25cb6fb4987c92dfb68683fe97cdf3bedde0260786" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6739c51b2034bca610c777060ee327caf1c7760553986e333bb2d4c075603180" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.12.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5d79db632b543dd049173d0902710dbbff83e26b5bbe99dbccb5590b93a2dbc8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="89c21d3e2c6a6ca572aa17ece3a316ddc61a2ace05497941c30c4100e5ab9879" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ee15da406984d3ccfa70a9e597e28c9ad7cac924045d3e5ea76f14b9c7be95b3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="ee6833febfdce400159a303b7e2d24900d0a4fb0472dfee63ff2d66d83eaa581" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="6519f1eab1ee9cc73b9756cce32baa0095d6306f79f34b61a0d0be3d39e2c8c4" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6db5a9ae80c287c729457863a7170d000bdbd099d6bd5f8e129bb8f5723b67dd" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="51d0006c3a5438fdb13c9d8eb4a3e9732035e01adfff7b9b5e9a64aa08435996" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="54e3ea36b3940d4322244909a3d1f82a66d9a9dbf6ee04e03681dd7eeaeb4c6c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d9b7985216151b1635885e8da4c226dd3b769439fc77ec75963279955ab2edf8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.17.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f8973a8ff6a4f18b4e1928b384dc2d7b545a7316d25430686a4ed0f13f769ab8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="428a49edbabf8eecbe29ab754d4ad2ca41bcb8612005d43a6712e65e5f0be982" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.18.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="4a7455e07545e6e0181cf58c8c24360ff5d46c0c8e0118d9d413fc0ed22caa92" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="af5ea035acebb2ebefc48f6a6881c3a9bbd2e5a9c86a56a301f6899b5ec7581b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="d70188eefa731f33b0a6d94f290117518750ce9987a3f21785a4bf78041633f5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.20.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="b786faf67f6541d894e440883796f9233adf55643febac78ec91f1102c194662" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.20.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="fd55ebe9ed001b9dc561fd76f87e8351faed08630283ccebc64b55cab7b8b007" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="2e82e96b966fa6688148c0128c1770d81bc86ce8adf6db9dbc0c7042a86cdc74" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="c7e5db81717c8e3ebf669933b66c3947e30c77fa064d6791fcc91f21f7687794" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="d9bee66b79ae02532607d88019137824f8269b0299cd42dbdbd4b911088fa1ce" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="018bae0b6bc00c13004f22b2a1febda8c298eb7e094b2d3d2b52ffd7f4532dec" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="9e1d526777961b18b186eb0e7822d7176486691c9df0c087e681ec05b90426b9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="5868619980fd3d9c4b59b47650b78021822c1583ecf6177776df342ba6078bc0" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="a11c483c5aca61df263ce006f8d20000e65f8ee585c67b38688e8ec9a544f065" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.24.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="5688dfcdc7bdfdc89768f0979912f909d75551f273a33707c0d214e266f0bcc2" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="da36266d317ef06023b02e6e61caacf38b840bab5ad39903ae0dcfb5218a9338" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="eb72ab3a284ee39b0411aa2b72afa4cd03046e29d19a5a512c8d4d058df70a85" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="e2b8a004fe2d9d7fa405496d74dae623414b6b1699e57683967c9e8671ac44c2" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="18.26.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-
- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
-  asterisk.c: Add option to restrict shell access from remote consoles.
-  res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="f8a4254ffb3bc553e2baac98a1cb603a47e8fea35dc2a76e01196d25da7e2422" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c676f0575058f404c98cad08e08bc5ed46b98d72a7231a87dda3d8aea00b951c" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="702bf1f1b5d78a026d8dfbf81f9f7bc273f426c4d6eef29bcc9fd1214660e385" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="682e0bc09a4f117905dac187323c47cae35fb2e1d5d760449a1a11dd8e8a0ee5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2763bc452ae32666d127bfd02bb748f503734416369cd9850831aba3c45a8c81" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="87f62d032e0c025025828456d39c156c13a5825e5be861878863769b4d44337b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bdfd64303568c93d027b81a8b267a7a495ac561c55a7c707cff61ee040439bd9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="0271bc95e24e21fb59fa2899e397859203233ad5b6c0b57da6be88ae4b025925" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a0301fdc5f80709362dd489b2b444ca204a51fff1a2770c7b84fe044ed8c9b28" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.3.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="62947372ff106ca225c53f303de8ee1b623c913f6204f5e81641311875d22bc3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="65f6415223af6b145210fb6427f8f4ce9a6e3202a9c0c5e1200460ae7995ab37" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c55eed8413d13781cc03f40527cc60ae78146981f7df8f79465a73b348ef0809" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="edb0c8de3a8061803e1ec201f7060850bf7929cec7af83fe6fa694f827edbb56" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a72a346599dc58117adac034a8dc48389370783363f3ed732d00a8b7ab4874b5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="0c1529b2e6b1b658eb743cdc0a432d446b220a0641d4bf07c4d2e54abe3dc23f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="5c9d4bb53416a4838f4715c6d4575fee27867bb94a1d2c6a4c2868cf1d8a0732" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="44dffb753c75358b2fa3c59afd55b5bec4618ecb4f3f54389b5cbb8e62b84acf" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5451de54c0d4a085fb48ee1bd0c563f7501dce2ac241939e509e4f8ddca98f7a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1f3694950369be3ac8520d39ea5f67fe06ad800269c0d9c8677cff5186fbd674" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="19.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="b3271ec0b779fcc67b8e4c2da8ad868a46f6eb1d76624779a7a9352cb5e6618f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f4454ac397d4e5f5e2068280d0c5109858d751f5be38db8474dd4e8141cd1e25" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="558f81f7ca19ca8acf7ebfba89301bd11447e182859a6b8ffe1da785273beea9" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5d4d78c13a898bcc60c2725dd693badf4d4b714e83441be246c34c69d56b4254" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="6d51b4bb78bff3e051d12cb0d6dec612116897ca398e40a5f6041072244550b2" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="009b819990545e479923540ce9aa88877825b49216242e5da8eddab480c97370" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="8f44047e391b06558f8f1c86e2df73af5804f9a859e6660d94534a03bf98de9b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="8ff4ec46423b898c9cb786e3c599fb968b6cc59a6a3fa02d93221440c5e5963d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="92e8f4f44c4e93112400414aa1c579a3069139e081cb3c139593fa6fe6cceacb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="e86b5a0ee79aba2209070eaa37ba0f39932214ed0a2dfee3cc5fd546d2a42dcc" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Upgrade Notes:
- --------------
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author:   George Joseph
-             Date:     2025-05-19
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-                       has been added that, when set, will prevent remote
-                       consoles from executing shell commands using the '!'
-                       prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author:   George Joseph
-                        Date:     2025-03-24
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="72e13361cc5fcb19ceebaa23fb5c17905f3552a4d5c62609c3766abf5487cb45" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b4c784c8f5691f007380f70edd4f0e32dcb33578ef5c37f3ce1d04e511a01a97" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fc17ecf661b31a096a24131bd8b3dd1a47ab54988147b1d71d8585322c6b95bd" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="98bcbb4c27be405bf9cadb1bdf5f2479420567b4d220105d824410fefbc91dab" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="ac6741fb1efe7cf5a4cf8c5c89788d543b56e1fb08cf7485cdb832a396be56c5" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="b86be7faf381da0e07be291f858946dd4e75c2afe756a5cb4dc11964184d2315" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="a7c5daf51caf7c8998f1cb61ad0aff70bff3b6d313697cd7faa00f8c8816876a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="689e72f9be6dde01d3893ac4b1dd69eb5415fa516b7c770bd024b75560eba7bc" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="d8ffbb5c4c58c40a5e1831d49bb5dc18f56198a2822d189c2e08f94447fd76ca" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="28588685f039fe5af366fda70885ac84063071cd1900419a8e4849948ef9ff09" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="4e47a21167aad3870bbe6ebb9a020a0773ac35ceb212639b07b1bf81e131cf53" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="39c325723c475183bd6e186937b7ce5084b035edd6900c1c0e0beaf6949426d2" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="32f10c314b34490bf324712607a4d70c723a818f6ef8b32fc7519a2432b0e3b3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="1e8b307951ac4c87ac2c89a3f7bd10a8afadc42dce3a8bc7e90a5f860de56e14" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="783ab14559902759a34270590a0a2b3ea0361f83a93e5c2ff5b041f9ea25b0da" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="51ecb003acad50a3b0a97c7b28de0b3829897106c5667e9ed48fe09813fb378a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="da4b3f0980bca8383f43b4cdebbd2f2a0d42859a576f7dd11c8c3d5e27de7c36" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="3a991ee0cc2205443390a0359b9fbd5de41cec4b83ac100587421fb2dd37914b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="430683184638ca30f21087e88eb6d1dddf830dd1c79821519686c7f25d2af962" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="57a18af4e86fa9cbf550460cff3ffe4884ee868184368c0905c109aafe33d471" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="ec60559eaac174c408ab34c1df594fe8183d663d0d342e0bfa268a1ed6e84fe3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="19b03e56d49d66cea55ca6ad9a62f881568ecb16e4bee523c785c8c3e4d94a1d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="cffb4878f63cee6ea41383080bc2727950364e133108a414d8a16b9c789aa919" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="feac4ebe5b18d8ed96db2ce15236c2a0b3d056f59bcb9c86ef2b85443df42eb6" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="b9e75272315a5326538026bb8541c0304eec1cc21d9c3b9099cfc4a15a3e76a1" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="c883afcc1a2c1ca35da79e9561652c19d114fbf763d5223633a5a296c3929d9d" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="9cac8d7411a5e8d4b14245aae2850b75f5b6dcf34549baca88ef6ad4a8b5b2cb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="af56d8ddd79db17ca4f314444cf204ec7d40f5ac35d3148ae26e1457f03f06d3" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="0d31d443c5b01fe2fc728d4f831f42aae2b4eb82958fdb7c3a774522ac83fc1b" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="bb14bb69aad067821ab626620555f9bf4dd74c98fd2e3c9189cae1e3ba53bd4f" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="0f3e0474e1dfc6c6d8f99cbc35dcf2e4f7e0fb43a391bd62c6e62b0ab0736d94" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="655bd369de1595b86dac2fa5b6ca9893b0e5dcaf49b8f7a2077a458097614c42" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="9ac598ad101c96463fbc1d0b888b435ba91e61da9eee45689a87c2df5d9ede5a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="9481a9ebba7e425a811527b9e0c4be26ea5c2a0114382a4f43fe54de3a14bdd8" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="8cadd60f9ba14014e2c4552e30a42d72e8e0a77a01f8057d562f5d3827e5fbde" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="64b54a90ea26bbf97dbfad2ec6d6f8a8347e97768d42f8db7dde6ca737f6c4eb" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="43d2ef182fc11655652638258395319723dbe22f763265f139f42ff11b52990a" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="a0049f25a05b4934676eb5c1c8bfbe1cca2d3897244e7e9c71e4eba2b7839f23" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="1c551949a06ae2e83eaecf31d71608731933612ba467ee05311b1434e8395dfd" name="asterisk-aeap" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="57a437333890e1e20aed0285f6ff040f8ec2de85a184ac8957f32df308e00e42" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.0.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 14.5.0-4" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7df3ecc385b595ac1f59f4a2f8cff9b1c3ef1e84afa2b8fb9e404231bfca29fa" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.1.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5c6e20d053af73c6174f13e70c96850b79f7dffc78d8d36f390c077759a9ed11" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.1.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="af27e892f09c0528596249cdadbe6ecca304744c5c3031f541d2720092eb90c7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.1.2" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267200">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267201">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267202">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="55fe72d8d22f982836b87af6172d2998e8ed61bef1f36abf8004ef9713ff67a8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.1.3" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267200">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267201">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b7a3096cda182bf24799e1219241a1ace70e8a840cd8ba8d177535a71274d69e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.1.4" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267200">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3a65d45239ef7b3c6b3fb0c52f7e51dccf7666452a25c3eb0967a7b1a690d679" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.1.5" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="05b1856e44e7ae6a7fb8dcda9eb53e48c7d584818c931cfddad3fa87d4e9d1d7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.2.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0e9f7bd765da1d92646c000a3fae6677fc60a739d38792670bc251880727c538" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.2.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6b627aec665d63ac7222fb8f560b384d314ca35ff88008ffb97853f6c2e1ce0a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.2.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
</package>
<package pkgid="af5635e3e88e15fe4b19bbc8581257f16603ff8508c170ed2dd1228006993be6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.2.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="32588281512ea93c40f1689dc40c9b530a0a3f8c51ceeac4739ecbdd7e1a9ba3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758404">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758405">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="027b5b7ce768aed60f32120d28e4d72f46f65e194dc4cfc62bd6b5b859925ed2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758403">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758404">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3b3b33726f65cf37c53d0baf7fbf31e373be69b4e9fb528731ce12f37dfa29b8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758402">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758403">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="08579f546d363b06e8b1f04b9641dfa314739980b8b146d58f0dacaf70d97b99" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758401">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7d76d58c75644656f911fc1fb02a0717cb09765251ed3005a99513fc1367bc4f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7766fe91362bad574a8d74e89d518a1ef395ad44f495e3aacb316991aa57b51e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0d53091180b3abcab2594e6c07f4aa5c227e85f80d62ae94f3f124f31362d42a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.6.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="155f7d749d6ddebc0cb6cfc3b81fbe9b13d3ca4a88b0dcad9b518776627168ba" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
</package>
<package pkgid="af60a53fe6eef0da7f5fe2bd05d3f618a0a19a9b99eebacf338612332fba22dc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b2b7c4923df9e31778dffc7c1bba9980ecde1c790078776bb39da2c8b422bd3e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.7.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ccb7c5dc60b1e5354a11476d774032f6638433bb5d15cd75d58bb91037b67dd4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.7.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f85ca4fec227ff1feca771c7b30d57a328690a254a63fc2eea36457e69a851d2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="15.7.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.4-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1,
- 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-004: Crash when negotiating for T.38 with a declined stream
- When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
- responds with a declined media stream a crash will then occur in Asterisk.
-
- AST-2019-005: Remote Crash Vulnerability in audio transcoding
- When audio frames are given to the audio transcoding support in Asterisk the
- number of samples are examined and as part of this a message is output to
- indicate that no samples are present. A change was done to suppress this
- message for a particular scenario in which the message was not relevant. This
- change assumed that information about the origin of a frame will always exist
- when in reality it may not.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="56074f37779f20cdb658e40cf6910d72a6abe8327a87465a4e6f771dd3f81633" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="13e6bfa6fbd9bdb4fa3ea04a0273126777211fee2995367be7e9e85763899ff4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b97ebf1c3ccbd6507cdb5852601cfaadc0905cd9456a2f9fdfbea4feda56f6b8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dd5e765ba094b4199a5e99d2b58d50da13dc3c7aaaf9d59372dd9b8544242370" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0df888187bcaaa1a2720aa7fbccca25a1033cd0ecdd3c556015e0a6baf34fd5a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.10.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7c54d3694cfb90fd9614f17cf6beccf7a6c4c40f3d986ac99593ac6e5e051db5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.11.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cdc182706dc3281f36117a6c579dfa5861dbc17b7c51f95de4cb2a906bbc5dc7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.11.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a07c72db9c36c78d631b56d8d427c64e9dd91383c3f471a5efd2f2bd2df6fad5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.12.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2a761317b42ced258ba6cfe886f3850f606cbc8bab661bd20a3af78fba9d1e7f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.13.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fea8c4a22df6789c1ea01eebb3d7c6a5bc3331e2cd716859658d55286aecbdb0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5e5987cb90778bf143c3740375dea66ad2163f5071f20c42b9768fb49886900b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="06442ab18495757abcd0a68a164e370d815ce53f5eb79147eab5185fa5f85e05" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="56bb6a52c78b0b5fdc587ada76412e13b71e0096c607cb24725d2bc3edff8524" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="74b85f723b47ece61cb52f3db2a14da8d02075137344473a3b141a74d58b82b3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c54d60a334eb19e339ad3799a99786c9e2f2dcdde59f8cd0d0372fe6a77fea96" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.16.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="437e92783d3ac9aa09bf0b49184c881dd16702b72571a9cc5a98b2963a00a80b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.16.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="96cb0bf0da49f62fcb353b219e93546f26c6d7899597c27e99b28f2e233b168e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="53664b1d8cf9ef1677528adf95bca72714f614248275b83e21b9424921fd8abe" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6a86e0b36dabb6b3938991a2480950b2ac2d79f8fa3e76e08bc40aa2e342c5e4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8d00a49d6d844099cf5f2da405a32ab1b3e4fbe58f91b9afe585e0dee0c5a669" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="78d2fe2d19c1c294a8e6eb69a70d09f9f2bd7e37626bb2a0baab3e04bb5d5bd8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="149d5c85682b7baeef7a987d0ad4092882f081d4ee5095128b9559b5df4df9b3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="276bae765bbc72ff47026f27e805859be333dcb8774c05ba58ca3ce4334c3656" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="38d49055e014c9d3fe1cc139d72d2530c89836830cae76ec7331161673d6b9ab" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="39c78b8f5161cf4de9f6c41bf61780c24e0273ce8b9b586a267a88ab25802f97" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="577abdaf66e7b9e286204fbf29747e2a8a7c1b491409a211643736a4e90584cc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.20.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="c43928400dfdfccc4492315f10d4d0fceec45bb56f04f8d612aa086f9634377b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="98caa8bd6eaa52044606e6a5d576727e433b9d6f5c66ee5e9bb24d7f895fc578" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4256547796da392fcfd6f97c43ec6846501cf145072cf50a13d0eb2dea3f891e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="13dc0973e63eb4d80cd91a7c86751aeb9429178c8713f74c9fc687e16df393c4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
</package>
<package pkgid="7da41b122d374ee7a1dcb2be142d4fa5aedd8b4cb7c16bb9331af4d91b362278" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="722b19abec7d4cad2167194fa3b0a3b4ad06391451e0dc86ae405aef87a3c41f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f63975efb1e6158f7ec370896f4865ac79bf0605a48c948da5f1a1c2734d4324" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="46d40e618db31116030b50a4f0b435c250ba29218cce57f949f4fa9d4ee3c0c0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="605a6be52f2adc878594548d8af41d380e653794bf856c56ff48c2e0d2d7500c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1585c2dceb4a57d5ac2f0c1a22ffdd8e5ef0a01e5e11a47e5f4d7725169482de" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0a642cdd2ec8ed424aa03848a0027ebb8d05553f5d2313a4dd400708c7ba9e70" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cc850a28e37e8d02f7ae0d48281fd8c6a150ed064d873eca15b0f74fede976a6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="17478468dd1c7e6b743003a1b3d5806870f27bcb4d833b4295a2b9fc97025ac6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.27.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="de537a150215a54fee47853744cc2463f50f75a20ac65f0caad946f2713e0e48" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a785fbce726742c82cd5644985f83b48dd824126c4742c3f15f953bb3820ce09" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="815e5726968b270a6fcb115e8b6554cc76d3fefa0f0e1b6b0f8a084bd476d6ca" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.28.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="04fe110967a27b51c535a2b05f42b746f87c464557b51624ec2289dfe7e4a43a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.29.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a59e921f2d0d3141a995a58810ab964b6f587765ad22ed73f3a6eb8f363c910a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.29.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="29e07e9e54809fb7d19192046e0cda7cd55785aba0e09345197f3a3f8f2032f7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="87d17c4c2011235afa5dc116b0c54391a45a8c959d37ac192d9976b9b8153165" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.30.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e7871df9d169b24dce7b4f6c1bc933c24a85ef1fdb28156dc69d12a25ae34a52" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.30.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 16.30.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/16.30.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 16.30.1
- ==============================
-
- Links:
- ------
-
-  [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.30.1.md)
-  [GitHub Diff](https://github.com/asterisk/asterisk/compare/16.30.0...16.30.1)
-  [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16.30.1.tar.gz)
-  [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
-  apply_patches: Use globbing instead of file/sort.
-  bundled_pjproject: Backport 2 SSL patches from upstream
-  bundled_pjproject: Backport security fixes from pjproject 2.13.1
-  apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
-              before applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="50b76aebf534b400b599834435ba796e7a83e9472ae37848fbca40b59e586046" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="668dc559437c9a9a33b89165d8706c91983fa0f27e610cea105e61bd10cff59b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f4092f794f5eb9e24005c471cf51466301e3e4a39ba9d710301eb0846e10c414" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2f97bc580da62318a688cd8e3b9854565dbf07a72ab458d4f97767169b3efe74" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d27df971492a8c4ab1d39fc54b99495641d9ab0fc007b99ab5ab12e667ae3fd4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="338ecb248336a4800481a9d26e3d043012b6d16c86e71377942cfb4f8f2973a4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
</package>
<package pkgid="fff394fc1bb7ed12babfedaf29d8be233ef81fba5f3f1cd648b7c0ce55302245" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-1" date="1571227200">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3f73ccfde2e35728b13d647a0e50a628f800cc8775ab5405bd394d87be69130c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.6.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-1" date="1571227200">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-2" date="1571832000">- Added res_json module</changelog>
</package>
<package pkgid="29a54e1dc366b255b3c9e1c15d2380efd7c3326b5348622a20d7f1adceae16f6" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.6.2" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1601853c067028d99b2d960a350627010e098c1dc94bf649504443fcacf82884" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.7.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2b1c4157bbf6f3e2856b9551e3502e346b3904037f30e10dbad28c875dd25c6b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.8.0" rel="1.el7_9"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="025103d00bc64ea0b1e396c77f9752a3f217d7e05adcfbf300b1e565c33dd75c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="16.9.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl) 
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes) 
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f5465b317a3175d0f54b08bcc9974bdbe9ba59cf10b1b9672ec135cb9d9c696d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.0.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e765af59f2f5b06ca0f75875ea26545d9775b1edbe408de9815a70c075d08b8a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.0.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d5e6a4ca470e8180af8d876a3d0886e90ceacc047e20a9e626f6664815ea5df2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.1.0" rel="1.el7_9"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="31c794ae85b6d4f02e8b3e2340a490b0084f429af9d3c7012d6c1904fde0abec" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.2.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="299a727e1403af6f8f2184c03582a7d9ed48991e2bc7846246bffd729f2947fb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.3.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f1bb88d34782f3a1a1714ec010d26e48b756f561f1481bcc7617be30e73d9973" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.4.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d45299a5df5484eab7cc4dc5d7b8ecec2915be357107dfd6d7cbaf5bac29d119" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.5.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c4732614230a5b8e9dfd2111e679e1344cf45c577b30e44640b2f259bdcc1349" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.5.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="650ad3a2d991d7e52aaccdd0096bf8853f7d0edbb0ee76fd5c5c94d237ad89e3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.6.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3275691d0417a0e9b163fa256ffabd81d8879de139498216ba74490eac271ad4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c6e8a0d1c1704653c9738a141f3aea903113c2b233f7ede53a52bb80fc74fb5c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4dbd7b64b6826e6bd11eeb826d07b1bb2975a3321dcef5abad5e361daa70c779" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5ed985a150cf87644755af45f056180dac499b6e71dd107504cb0cdf13a2fc3e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5a4557f11bd47376217723843e9a2fe337df84850605d7dfe3368d39e51aa8ae" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d46aadbf9d9e403dc04868b518e5d65aa0a742283d4f32d28546b32907fa3015" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="66bea87b140464f6507c9acd5140f034601e69b8bcde1a866c7399bf4c6038dc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="024e1e2d9313418a99af8522929205cb2e6e2fdcdacadab5fc5d65f1a6c7a5b5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f92d8d51ec82e46b973ec1316f45e34c3028a6dc6699fefc92df02dc37c450dd" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="f698004d7c742b142596bd4e1998923a1aeaed3316380c47a01f84bd071d9149" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="17.9.4" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="5dd82b4cc5416b93d26319683193c9409559bfcd82305e733e4a498647658847" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9b5fb94888dc66d52b5be814ad2f4a8a56845a390b5762856b5796146411ab22" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="250428c31da2b2af82d1dbf66aa04babe44c256024d993a96e26e5f6626fe14b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="71682a292b5965004df23637d6cde3cea5905fc00a24f726b6cfd73cfdbfac2e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9043932fa640571d53b837ceb1e710cdc646fa53e30c3d838dabe84591eee9af" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.10.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b96813aeeb6169031d6353196cde2fb8d0558f9661915150b9b3a517371822d7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.10.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e0b553e7c487a46434bd64d967e04af9c66009e7921e221d16a2d00a97c11f35" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ac8c5a2d2348f2a7882668350e3ddd186a9644bf90c255454aad4b98a7b3a658" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9a66d384333d1e69e6b4fa90b8d1e9ff203bc5c9a3632e9c509785f241481eb5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="8eb368fada9f3decf57ccb197878127627f413a314ffb3c91339d09fc2afe93b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="62124dd811efa3ec80d3286ed3ce6d666769afc99cf5cb522520003312c58c0c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.11.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b05571c148fb9e2b4a87185357fa70dde39c792baac256eb5341ef04a57023ec" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e3ef9d2bf501e62cf3be5ff182d57dee5bf4f909db44f58ef4fea775d3695a93" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.12.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cfc17258ff43d6e49d5bd96db4582f660d4fa8339d46ec184ac181367ddd2b52" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cb406ff8cc4e4fce9dacb3f49ac49fdd7f8de89993d604c8e8535981f4eec1ba" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="db6a1b0db3cea61f1e86bdde852d2f5e45acf9be43a4209431f898cd6e52cda5" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="07d3c7cc73cf00a0bd8770099c1ce839e49ca9a8fd85f1234700ad343833e8c8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.14.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="6c554f410d0d77a2deebac046c1c2f650774edfd5d263bcd1e1c953797ba7ab0" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="61a4cf74df846b21f56f8a54e5d64288c472737e55b938674d01e4304fdccacc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3f48d5de6b8e83b915d9469dfa92af71ebeda349a5389243a3c849c8871082ff" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="31ee5eba099ae3ec2c2406fdf5e5372ebabf4dee174727dab18a70258b4b7657" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d438e0a3fc39cc173e0af9e93f5a92195be12a194321c54394bd93e8df559e99" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.17.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cfde53715c7a74eb00e4d02eebfa39cc2714fe1c84d8cee042fcecb7d0d4ac72" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="91f56798561a61a31adeb81e4334e7e8956c6b9baccf724b058e9b35c7ff15fb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.18.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="a30f9c6db5cfdf11cc85a3469ceb5d43774f3167b97e4b9dfd2df8da2534d084" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="b0ac4662033612773fe005405366bd002311b16d676693f0939dad33163f2e86" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1fccc0aec923a901c08e17b13e430b20d555e4d4b82a6059429487f71e212559" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b921eb6e0b2e243107cad3ccbb47d548474811d969995c829512a090c7f0fc8e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.2.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="686a298d142b1f44475e67969cafa6b5d824e35c8f79debcbbd038a97cf2c2fe" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="652b721bb4f86325b67a8f3da10603c3e4299aaa521353a0b43588de7c5890ad" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.20.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="47c8c9b628599755556361b825d9e3bbb735f78a400b67fdb3df17a0818dc12d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.20.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="4627018bff60280988cfe25791d6c0ead448958dd7f87515a6527ed90d755419" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="9d66ca44ae14da6465b367d8aef084a985d7d596b4d078ce67d9a35a4bf71d8d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="ff085f1376eb6ac9c9012ace9324e1ee626921a5416a2777868dc84a1ce6320b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="84c32e7fb23ff4e5e871e133fe8e7a6db3476443fdf6999eebb75f10e5b21112" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="15e79e3604a5b8f85993ec58382073f6b682203ea958954dd9f282f1f481d9d7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="6f9e6835ebcf9153602ded0c8342e05092306ac6bb1054ac05ed74a445fc6cde" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="73ffeca124c39b1f313b26fb5e53d748571c3ebccb21e5410fd3c2b32c520d84" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="1dc479e27901a7c685dfeb3dc49d56c97ea343769a1f63a1e173efc36c05fb13" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.24.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="efe200652b7afaa6f250e9408ae6f8d5460389c477bb49c18a3f13869eaf8d41" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="04659bc7d3d44ede16a8070317ddcf8568977174da4f797a6e4fed385e851d88" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="c37c167d41bad69978bfbf036e271774510170cb8e1382a2ba56ee0515aa76e7" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="ba7c2c1bc4960a14362a9db05e8566e1c65cba30a64bc6382aaa2bc707ee3a93" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.26.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-
- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
-  asterisk.c: Add option to restrict shell access from remote consoles.
-  res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="f1de3a6c53dd22a63bde274fb925386e57a2af8f3887e6093d53ee028b1f7dc8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="538fb1a72dd73d579740c790bac4cc53b3e7d1dd77976f90c9b0c4d20ba36513" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f6266603cabaed72f65a1b49c7443583e9b94d4f9241c3128181bed636d83366" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="938c288258fc8cc887425fa84c0da37ec9bdaf4813ad77e71a781e8886329d55" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6fe154ba67a50524370540ef29d4ec38e16801ec2a87686544df41f87f972e76" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="b966867d97af6fe1e80c032e44c1d3d4d0137bd1e57e31284c1f427ef2ce6a00" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.5.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="09ff31bff7ffe2301fe695c887eaaa8a8fc95755381d8b6cd693b29a34a04fd1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f29f0b2653306c5aa2a24368df490d283784833874ac8052500f8b71de7d601e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="2f9a7e4566b5e4b169c42d6ba539a230e4852bc24716e71c43bdef10056ab165" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e7e595efc64cbe944918a47ce5e28cc5b237b34eeed34389717759a235432632" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d793403b9801bcc8f1a81dd996169dce9ec740dcf5a7bd192295d12cf7c9cadd" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="053c900aaa49e3ed94e6d80742f2b6f28d2eaaf47fbb447d84922a188d0da080" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="18.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="45fa22bee53c50807fd9b5b8b0833e913fd99730625439baf3b7044605b25e5e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2dd7f6762b89c5c0398403ea2cf6d696b4cd543bcd5927c439e2d37c4f224422" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="58ab73faca99c7d8d8ff76f2f3b33ba890da71616a678459f1c875c7fad4bf10" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ee73ba582377ea286710571dfeea75a34a83f2a00cb165e96a8fedc5ef773794" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="67a4e8763cfe196a0f847eb78d0588b07daa05289df06d83321e028da4a552fa" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f83e58ffd2327223f2a3778c17313e782bfa19187102d9eabb7c03c507a67d69" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3177b5c033545434edcd4c98e23e4544c276c13e0a584bd8811edcb472cd1518" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="497f6a5622728ac5ede873c8b513d0ee4299a7cd007834f7e2e8aced2c33c83d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="39e6bbccdac35189840fd447fd16ea9cb3eb1776fc597d27a15f2b1d610292a8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.3.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8d45f264c3bbc5d136ce0ce8a5a4a86f20c88c3331a62534684b143966424f38" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="516295944742b859a54d9cd36bec8dbed00a8f3b196b39b29217d42805f6851e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="67c1b5d55ef5be0f130e6ac98166992e40f37aa47ccad913fef4af0aac58de6a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1d26843bf49a8ff9a609ba81babd70fce42b65cb4579e43d404f0d59b90b3fbf" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="987fa4ca7ab4817b4a46ae4e5c7ca1448333386183f99b9370194e0570e160ca" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="3913a1e01869fe7a6864892d8c50eef2fb7b7d8d6d806ba4e40c764d0a7cc0e3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.6.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="2f8fafa3500fd1b33e175a18a04ce1f153e5e2044f766ea3390b5b52dfde66ef" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dc50d480d66644890bda3bda134869b257b6ce0b39f65e835db30e4f5c0a9780" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c5ebbba0d50fcabd9bbd134e7673ddc3037b9d2ee1d2d70770c1a7f55daa5d7c" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3fcea30db49c02dee86c1dff0a4e44e4c9dcf02808ad299d703cd7f124a68f92" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="19.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="db98327bf6291952d4a34a4e2fd19a24a925106e08310f93f715c9e36c29b6d1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="61cf817bead76700f11320c32a7dcd79d177c3ae18abd11352f2953a5833aa36" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a59b5dcb3c7c5dc5cdab980f1d176f12ee04ad393052d00281c738504997d886" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8e8236177ce62fdd9acb157d55cb2d8010b2e113560aa9621ccee975ae8617f8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.10.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="f59660b4753449b4f3bad3c843dff0defddb1ca6ed130cc6fa81be798d034a71" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="1c122169bdc31754080082e0d2374cd02c28a7b4b59af6b1b50e6a419180a6ad" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.11.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="dda36b7c81573087bc41d99a35d02b11cde4413694075a52a77ea69c883e86d3" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="882a1cf30531ba14eced5c55e3f20c93cbf0ebf7995904b866d0a92addcf1229" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="26e78bc2b884e27e979e059c2a4654a0a3fee2847efb3c4ce229ba3561a2fd06" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="bfbcaafee2a109a1938ddc0268b05607cdac12e445eed7b9f1d77db155de1a6a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Upgrade Notes:
- --------------
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author:   George Joseph
-             Date:     2025-05-19
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-                       has been added that, when set, will prevent remote
-                       consoles from executing shell commands using the '!'
-                       prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author:   George Joseph
-                        Date:     2025-03-24
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="e6268935ccba237749ab63f407c17175a3c7699deed77ba382cb23cffb60f56e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="122a4aaac319c6b06fecd962bc7097a95e8a4642251bbb9ebd9d06d5ef8769ee" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cebc689c7584306e5b4d503953f86a3dcbb129ddc48c50434a3938fc928c7f43" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="377d151bcfb7b632d2dfbfcd1523f7049e0aef505b6042e4855e2abe3322e5a1" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="b08f0541268076f4aece8217bf89cfe4574497a616c6e331add5d6753a6e95ee" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="1ca7c176f57abf99d586e26c9105d394225ac5cde65969d833310fd946a2f69f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="66bf2c6e29eef7d63455a7b3b6d6cdc9bdc60c688483fa0cc5507c474a25bebb" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="bf258a92ef3503b07b785aa878f9f6b5a9da741b19cc267c2dae540ad8134b30" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.5.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="918bf7a7dc3262aa8d9f59db2bacc2929dcd638ca1b7b193bbdf80eb46a339f2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="8474d5b662e2a88f2ffca852719fe445bbc57dda461e00688a1383050dda5376" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="53450bff051f732010519f61375de4282b8903aefb5862e710211708f071cbfe" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="778d2780289124523cd1f039cf1fe309d13b5c0fb158459f1ad9bdf1c9d5f4dc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="3636919fa5ba6acf88afe8293bd33d3266b8862f8c550bc76061976548e3747a" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="95a863d675099d203217017279d80efc5da0af62e051f842838fb4c5d56f5466" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="063c8eca71fedf20eed8ef021813bfc3eb05d5fc45d727d5f92df7bfa8c8ed19" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="b7c941f1dbe6d713c99a442b049b1112f8e055b077a31287b289eacea6c09ec4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="20.9.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="6ebab50bff3997f11d206b383678614dc841cb2f8983cb58d8d175b1c889dd0e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="6d449d8087d2a629a7b3052ca058531397b6e2e8a345ea8850676b68756d479e" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="afb23b55f295aafc515f17585e40253dc9b1aab9b91e72579f687ea0d722e11b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.0.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="9fbce6fc63b93ffd050b8f357cbd5322aeddd055ab6e39b54d22980a5dc11674" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="70330e38830b1b137103cf1df290c0fa09352380cccfa41eb9fb9f8922bc3d81" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="021b45b10e60bd948f7f2508c67e68d89892e1abfcdb2d04136f01bdf2447be4" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="547ad0a72f91380fd0fba059931ef4df3ce4a759ab7c1d49b8bd1558d2304536" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="58a9a1ec0dbbfbcef431f3140c58ae3acda14c9b9b6e53a8bede9df76cd87d94" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="5ef0ce9ff9ee3a9aebbbe39ee488cd3e1ddfd7ce4897bbb2d4869775e4f6c5e2" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="45a972f33e59763f06947ea2309cb3a77ab6e58638bd84f7cb37a63142bc921b" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="ff9bc2cf57212918f1e7f017c91d8269253c0fcbc9aa7dad47533cf4ade97201" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.4.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="d1c92c3c44db2ee64d152c66fec04da80e4fad8dc3346bfe83c4ffd9d0a2949d" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="a5e3cfc406b5af67045002fbf627409bea6ea07ab763718d7dbbd115133b66c8" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="8b3b2bf4363d035105ba54476096857037f7e7a4c6388a5a57d8f7bd320e73dc" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="5c1fe5e6dc8d1f66501200d365b03ec42c673047f57aaeaa3affbb6cf3044569" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="bba054b4029550d5fd27041a0f6c6b32e531595e66a73800fd7b707adcb10175" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="bc3b465a43026a2764235722695761be938a5559449dbf159c8c805a64048daa" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="21.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="f95254a1328888bec247eb3e7ac22b41f05dcab9edacf408cedf1a44efad1e00" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="9011d4919428a1fbe130e323b77dc83bb1fce41c6fe119c2e88b305d19f046be" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="61f59808a5fe1429f29293230b73196e8ddffe9086252f18d5339bbf36594226" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="8066fbb0d3ea44d7e547d809024e1db49f361526d7dcb6ebc4cbc04ef528d309" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="076fe29930a515cf246a8e172642553a44dec248c90ec70da33a33065516999f" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="fbb3f9a94395071aedde9eaa7f0f7b153a979590e0ec729604ebbaff91e550df" name="asterisk-ael" arch="x86_64">
  <version epoch="0" ver="22.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="fe98a15839114f650aec5884e1637dd4eb679ca76fffb255ddd3d85448e87aa8" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.0.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 14.5.0-4" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9c5da5a95605ab2c19b9f88e8776bd40a1675c9ce17938a0f55385009a923614" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.1.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="24b62a348acbf5958ee71e5bcd23f45ee28862336767e664e83f9a9bae80f212" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.1.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3033c8a4b9039b97e6a336f17db1ae19a8ac8885a60ead60cba13d6582f95be8" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.1.2" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267200">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267201">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267202">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5339e7faee0151106fabf9b1f7098eaa74748fcc628de7f09f9b5ef380424285" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.1.3" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267200">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267201">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bfc71758ed567bb4ecf9c97c34ac3c1eb34335cd1b14ac6a7a96d40b8d5eee84" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.1.4" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267200">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="494b9798ffdbf9799ad829091be383fe4525b45673d363a86630c41e5c28bbbe" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.1.5" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ebd30db922ab082a922613aa4596c72d72c56fe0f9e29bc39022a7fce72fb584" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.2.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a42bc71975fb7615c31f34aeef915c0196a6904e0691623c500e39a778873a46" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.2.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7365d9284b3bd35870f91bb830a2f13b3f77587e0dd7819a345a5b239f519920" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.2.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
</package>
<package pkgid="67053d679df5d41b72c025220eabfdfad3b6d3629347847701f338f431fa8165" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.2.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7be1d39a1577f3ed675b2a0c313b55e187311ef261f584fa7f8b7a5e7e8fe7e7" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758404">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758405">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8bf5b7193da8781cfca87591327fe4c59be843281d1dd425f1e7f709069036cc" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758403">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758404">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c921de2ebd1f9e47013f630b18308a030acfd32eb5b1ecd813e6421029a223e8" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758402">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758403">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d8cf1308d8a0fdf3769306de9bad47c7811d4aa19e9e6a40d99220f04f6341cb" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758401">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9f95ebd7f8059f351bffdec2fdc14ec609146e21ddfe118411008358a3d87367" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c169fedfb2a703fc0858c07f13b15f423ae1585770a828f30eec4c40f4cb6462" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ede2c3e7cca5338b60a2e76122b7c4bd6e6f5673c37533d0a38618e7451e46c8" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.6.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="762cead9bb22901c851820f06782acff1d30e0ef17767081d9943606fbb9f4d1" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
</package>
<package pkgid="23fc678149aa2239402bd64092040f44ae4a0c60f94b2f565fff9bb5250721f5" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6d8d59e319a5219ee3b4cdc0ed4f83890413e68bcfda5cac0619fed14db5ad2f" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.7.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="75158759c41480470adfd02c3b11d958e3e27e2c95bbe23fc7ac66f7d82ff52d" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.7.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="77d668f374c89cea229ca48ef7246ed5e4876854c2dde346d25470de3a3bb896" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="15.7.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.4-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1,
- 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-004: Crash when negotiating for T.38 with a declined stream
- When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
- responds with a declined media stream a crash will then occur in Asterisk.
-
- AST-2019-005: Remote Crash Vulnerability in audio transcoding
- When audio frames are given to the audio transcoding support in Asterisk the
- number of samples are examined and as part of this a message is output to
- indicate that no samples are present. A change was done to suppress this
- message for a particular scenario in which the message was not relevant. This
- change assumed that information about the origin of a frame will always exist
- when in reality it may not.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="00695f99c42820de0cccee68ce70844c2aaa3097a1fe2231a1e4126304077124" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="551023700db1a49ac88477b3f8b74b6179155bc7dce2e8eee51aca8fc3451550" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c67f0e17b2b0f1252fc512da518705258d4e02f9dba02819d1767ba8ba8f1e45" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ce3edc6ec069ad40dbc0de13566d41ebf3633ee1bbe3c4b221fbcf754ca9084e" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="060db37515d86ee655a4c4af75f231b4075902a5f782f9cdcf2e3aafa5e34b64" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.10.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fe083980590128d6de1f7f61f33b2cf3936f6a6b76f50f7b48ad6e5e8eb2c424" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.11.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f2fb10f4d3a1213ebd50d16d35ce8263563918ff317f6c2b55c0f26ec9728ef0" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.11.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c1ab91f89aacfc9ba294fdb669c1a842d712e3e01d40194c4fdb6abcb86c519e" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.12.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cb744b5995d94d006906faa29946483eeaa3da770fbac1e5dbc191436d99a135" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.13.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2360eff70c08968d2ab9c46b2f2349d26b8c8fb305f2eb7b11f46089460a2a6f" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6669032d54499222ed0af2beff24e200a1170c32b156324d9493f292aa8e491e" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4c7cf20f74fe049140c6107006ea2c93f5586f49ab8e5f8ad9fb2e2a5fd5237c" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6acb3767db7544a51128133ab7fc8b673c7b649a0f671d4fee43b55417f430f9" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="43e55f9bfaa462aac569d5c74b1cc94d830f9d7491581232dda178d3e2d78e95" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b013f3250106a46237c439c876e29b09f533e089fd1c5bed0ec653c9ffcbda9f" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.16.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6e18e2ee783542eb7080347ac890a7a2947459e85c5cc43f04f744f0e71701d1" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.16.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cadd30d2972332c72ea39778f2c797189c5192b84a195e8f201638b060a85035" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a06adcbdd43e04df5cbdfe846b3f7d10ace911f6da22fd2c5364bb4cb1a3c197" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6e03b8d084d125323f6a37272fa657bceaaa8a990fe15c1959f3ea043c281740" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cc234ab0b6123e2592b001be23fc92ce7fd809dcccce3c6895f2c79776511187" name="asterisk-alembic" arch="x86_64">
  <version epoch="0" ver="16.19.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to 